JP2009049818A - Ip telephone system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an IP (Internet Protcol) telephone system which can be used continuously for a long period of time by reducing the power consumption of a speech apparatus. <P>SOLUTION: The IP telephone system, which has: a plurality of communication terminals T1 and T2 capable of communicating with each other through a network; and a server Sv that performs connection start processing when the communication terminals communicate with each other and controls communication data transmitted to and received from the communication terminals through RTP for communication between the communication terminals, is provided with a transmission/reception data control means of performing control to alternate a transmission interruption mode wherein transmission of communication data is interrupted with a transmission restart mode wherein the transmission of the communication data is restarted as predetermined commands are input in a state wherein the connection start processing is executed to establish a communication connection between the communication terminals and RTP-packetized communication data are transmitted and received between the communication terminals and server. <P>COPYRIGHT: (C)2009,JPO&INPIT

Description

  The present invention relates to an IP telephone system in which power consumption of a communication terminal is reduced in a communication technique using an IP (Internet Protocol) network.

  2. Description of the Related Art Conventionally, a technology for communicating various audio data such as music such as BGM (Back Ground Music) and human speech using an IP network (IP network) such as the Internet or an intranet is known (for example, Patent Document 1). It has been. Such voice communication technology has been put into practical use, for example, as a communication terminal using VoIP (Voice over Internet Protocol) (various call devices such as so-called IP phones).

  As an example of practical use, when IP phones (communication terminals) communicate with each other, analog audio data is digitally converted at the transmission source, and the digital data is framed and divided into small units. By adding the header information of the transmission destination for each divided data, the audio data is sequentially transmitted to the transmission destination via the IP network in a state of being divided into small units called RTP (Real-time Transport Protocol) packets, for example. (Wireless communication). Then, at the transmission destination, a call is performed between the transmission source and the transmission destination by restoring each received packet to the original voice data and reproducing it.

Also, in the voice communication technology using an IP phone (communication terminal), the IP phone is wireless LAN-connected and the characteristics of the IP network do not affect the distance between the source and destination and the communication time. It is possible to perform voice communication at a relatively low cost. In addition, the voice communication technology can be used as a collaboration tool by combining an IP phone with a soft phone such as Skype or another fixed phone.
JP-A-2005-45737

  However, the current IP telephone (communication terminal) has a problem that it cannot be used continuously for a long time. This is due to the fact that the battery (battery) capacity is consumed at an early stage because processing with high power consumption is performed during IP telephone communication (call). Considering the power consumption of the IP phone, (1) digitization of voice data, (2) restoration and reproduction of voice data from the packet, and (3) wireless communication of the packet to the IP network. The battery (battery) power consumption is increasing.

  Accordingly, an object of the present invention is to provide an IP telephone system that enables continuous use for a long time by reducing the power consumption of a communication terminal.

  In order to achieve such an object, the first invention executes a connection start process when communicating between a plurality of communication terminals capable of communicating with each other via a network and the communication terminals. An IP telephone system having a server for controlling communication data transmitted / received to / from the communication terminal based on RTP during communication between the communication terminals, wherein the connection start process is executed, When a communication connection between communication terminals is established and the communication data converted into RTP packets is transmitted and received between the communication terminal and the server, the communication data is transmitted by inputting a predetermined command. Transmission / reception data control means for performing control for mutual switching between a transmission interruption mode for interrupting transmission and a transmission resumption mode for resuming transmission of the communication data.

  According to the first invention, switching control between the transmission interruption mode and the transmission resumption mode is performed by inputting a predetermined command, thereby reducing, for example, a process that consumes a large amount of power, and thereby a battery (battery) of the communication terminal. Power consumption can be reduced. As a result, the communication terminal can be used continuously over a long period of time.

  In the second aspect of the present invention, when the command for switching the mode is input to the communication terminal as the transmission / reception data control means, the transmission interruption mode and the transmission resumption mode of the communication data are mutually switched. A transmission / reception data control unit for performing switching control on the server is provided, and a notification command to the effect that the mode switching of the communication terminal has been performed is input to the server as the transmission / reception data control means. Thus, a transmission / reception data control function for controlling the communication terminal to mutually switch between the transmission interruption mode and the transmission resumption mode of the communication data is added.

  According to the second invention, not only the transmission process when transmitting communication data from the communication terminal to the server, but also the reception process of the communication data transmitted from the server to the communication terminal can be reduced. Low power consumption can be further improved.

  In a third aspect, the notification command is transmitted from the communication terminal to the server as an RTCP extended event.

  According to the third aspect of the present invention, it is possible to easily switch between the transmission interruption mode and the transmission resumption mode without being affected by a normal data stream (that is, an audio data processing sequence).

  According to the present invention, it is possible to realize an IP telephone system that can reduce power consumption of a communication terminal and enable continuous use for a long time.

Hereinafter, an IP telephone system according to an embodiment of the present invention will be described.
The IP telephone system according to the present embodiment uses a communication terminal using the Internet communication technology and a server having a function of communicating (wireless communication) between communication terminals according to a predetermined communication protocol (protocol) via the IP network. Are constructed as communication systems connected to each other. The IP network is a general term for the Internet or a network having the same mechanism as the Internet, and the communication terminal can be connected to a wireless local area network (LAN) or a wired LAN laid in the IP network.

  Further, the server has a function of realizing call setting of a communication terminal in conformity with a communication form such as SIP (Session Initiation Protocol) or P2P (Peer to Peer). In this case, as a communication terminal, for example, an IP phone, an intercom functioning as an IP phone, or a personal computer (desktop type, notebook type, etc.) H (FIG. 1 (a)) installed with predetermined application software and functioning as a soft phone. For example, a communication device such as PDA (Personal Digital Assistants) or a data communication device can be applied.

  FIGS. 1A and 1B show a configuration example of an IP telephone system as described above. In such an IP telephone system, a plurality (two in the drawing) of users P1 and P2 are carried. A wireless communication terminal (for example, a communication device such as a wireless IP telephone terminal) T1, T2 and wireless communication with each of the communication devices T1, T2 to connect the communication devices T1, T2 to an IP network (not shown) A plurality of (for example, two access points) relay devices that relay data transmitted and received between the call devices T1 and T2 and the server Sv during a call with each of the call devices T1 and T2. ) 1AP, 2AP and a server Sv connected to each of the relay devices 1AP, 2AP via the IP network.

  Note that the relay devices 1AP and 2AP have a function of relaying the data according to the traffic situation of the IP network when making a call with each of the call devices T1 and T2. It is possible to prevent traffic from being concentrated on the IP network and to distribute the load on the server Sv and the IP network. At that time, instead of providing such a data relay function to the relay devices 1AP and 2AP, a data relay server (load distribution server) (not shown) may be separately provided to perform similar data relay, Sv may have a data relay function.

  Here, the communication devices T1 and T2 perform wireless communication with each of the relay devices 1AP and 2AP at regular intervals (for example, once every few seconds, once every minute) in order to ensure the mobility. It is set to go and repeat location registration of its own terminal. In wireless communication between each of the communication devices T1 and T2 and each of the relay devices 1AP and 2AP, various signals are transmitted and received according to standards such as IEEE802.11a, IEEE802.11b, and IEEE802.11g. At this time, when each relay device 1AP, 2AP receives a radio signal from each communication device T1, T2, the radio field intensity is measured for each relay device 1AP, 2AP, and the measurement result is sent to each relay device 1AP, 2AP. Via the server Sv.

  FIG. 2 shows an example of the internal configuration of such communication devices T1 and T2. The communication devices T1 and T2 convert the sound into an electric signal and the microphone 8 that converts the electric signal into sound. The analog signal output from the speaker 10 and the microphone 8 is converted into a digital signal, the digital signal is converted into an analog signal, and the voice processing circuit 12 that drives the speaker 10 with the analog signal is generated and transmitted. Wireless communication between the user agent client unit 14 that receives the response, the user agent server unit 16 that receives and processes the SIP request, and generates a response, and the relay apparatuses 1AP and 2AP, A wireless LAN interface 18 for transmitting and receiving various data and a wireless chip 19 having an RF-ID are provided.

  In this case, each of the communication devices T1 and T2 is a server Sv having a function as a SIP server via the relay devices 1AP and 2AP in the communicable range (communication area) E1 and E2 (FIG. 1A). Can communicate with each other (wireless communication), and can communicate with each other as users of extension telephones. Further, by providing the telephone devices T1 and T2 with the SIP phone function, one-to-many calls (for example, simultaneous notification from the manager to all users) and many-to-many calls (for example, multipoints where all users participate) It is also possible to execute (conference). Furthermore, the server Sv having a function as a SIP server is assigned to a plurality of communication areas by connecting to a server (not shown) in another area (such as another floor in the same building or a remote branch). It is also possible to communicate between the communication devices.

  FIG. 3 shows an example of the internal configuration of each of the relay devices 1AP and 2AP. The relay devices 1AP and 2AP are communication devices T1 and T2 that are wireless LAN terminals (wireless IP telephone terminals) as described above. Has a function of relaying radio waves for connecting to the server Sv. As a result, all the communication devices T1, T2 within the communicable range (communication area) E1, E2 of each relay device 1AP, 2AP, and the communication devices T1, T2 and the server Sv It becomes possible to connect to each other via a communication system (network).

As shown in FIG. 3, the relay devices 1AP and 2AP automatically measure the intensity of the radio wave received from the wireless chip 19 having the RF-ID provided in each of the communication devices T1 and T2, and the measurement result is The radio field intensity measurement unit 20 that acquires and stores the radio field intensity measurement data corresponding to the information indicating the state of the communication apparatuses T1 and T2 and transmits the stored radio field intensity measurement data to the server Sv, and each of the communication apparatuses T1 and T2 A wireless communication unit 22 for performing wireless communication with each other, a calculation unit 24 connected to the wireless communication unit 22, and a network communication control unit 26 connected to the calculation unit 24.
Here, the wireless communication unit 22 transmits / receives data (wireless communication) according to a modulation / demodulation function for transmitting / receiving data (wireless communication) to / from each of the communication devices T1 and T2, and a predetermined communication protocol (protocol). For example, an RF (Radio Frequency) unit.

  The arithmetic unit 24 has a function of causing the CPU 28 to execute predetermined processing according to a program stored in the ROM 32 using the RAM 30 as a work area. Here, the ROM 32 is provided with, for example, two banks B1 and B2, and at least one of the banks (for example, B1) stores a program for functioning as the relay devices 1AP and 2AP. ing. Program data is exchanged among the CPU 28, the RAM 30 and the ROM 32 via the bus 34, and the bus 34 is connected to the server Sv via the network communication control unit 26. In this case, the relay devices 1AP and 2AP are realized by the CPU 28 executing the program stored in the bank B1 of the ROM 32 using the RAM 30 as a work area. Further, for example, when trouble occurs in the bank B1 or when the program is updated, the CPU 28 may be operated by switching to the spare bank B2 so as not to stop the communication.

  The server Sv is a SIP entity having a function as a SIP server (a function for processing a SIP request), and resolves the next transfer destination of the SIP request from the telephone devices T1 and T2 operating as a user agent. , The proxy server that forwards the request, the next forwarding destination of the SIP request, the redirect server that sends the request as a response, the REGISTER request is received, and the contact addresses of the telephone devices T1 and T2 that are user agents are It not only functions as a user agent server including the server to be registered, but also functions as a user agent client.

  FIG. 4 shows an example of the internal configuration of such a server Sv. The server Sv includes a call device position specifying unit 36, a call device position storage unit 38, a call target specifying unit 41, and a specification. And a communication device reporting unit 43. The server Sv is configured as an information processing apparatus such as a computer or a workstation, and stores a program for executing various functions described below in a ROM 44 which is an internal memory of the server Sv. And executed by the CPU 48. In this case, the program is not necessarily stored in the ROM 44, and may be provided from an external device such as an ASP (Application Service Provider).

  The call device position specifying unit 36 specifies the positions of the call devices T1 and T2 based on the radio field intensity measurement data received from the call devices T1 and T2 via the relay devices 1AP and 2AP. At this time, since the positions of the communication devices T1 and T2 change according to the movement of the users P1 and P2 (FIG. 1) carrying the communication devices, the communication device position specifying unit 36 receives the radio field intensity measurement data every time it receives the radio field intensity measurement data. Update the positions of the communication devices T1, T2 to the latest ones.

  The positions of the communication devices T1 and T2 do not necessarily have to be set to one point as in the coordinates. For example, the communication range (communication area) E1 and E2 of each relay device 1AP and 2AP is divided into a plurality of small areas. The positions of the communication devices T1 and T2 corresponding to each small area may be specified. Also, considering the communication traffic, the measurement data is transmitted from the relay devices 1AP and 2AP to the server Sv only when the radio field intensity measurement data measured by the relay devices 1AP and 2AP is changed. The server Sv may be configured to specify and update the positions of the communication devices T1 and T2.

  The call device position storage unit 38 stores the positions of the call devices T1 and T2 specified by the call device position specifying unit 36, and is updated to the latest one each time radio wave intensity measurement data is received. Memorize the position of T2.

  The call target specifying unit 41 operates the call device T1 carried by the calling user (for example, the user P1) and calls for calling the call device T2 carried by the called user (for example, the user P2). When a signal is transmitted from the call device T1, call device T2 (call call device) information corresponding to the call signal is extracted, and the position of the call device T2 stored in the call device position storage unit 38 is specified. .

The specific call device notification unit 43 makes a call to the call device T2 based on the location information of the call device (calling call device T2) specified by the call target specifying unit 41. As a result, a connection start process for attempting a call connection between the calling side call device T1 and the called side call device T2 is executed.
As an example, the specific call device reporting unit 43 is provided with a function as a caller terminal (user / agent / client) of a SIP phone (IP phone), and the specific call device reporting unit 43 and each of the call devices T1, T2 Assume that session management is performed between In this case, the destination information (for example, URI (Uniform Resource Identifier), telephone number) of each of the telephone devices T1 and T2 is recorded in the specific telephone device notification unit 43 in advance, based on the corresponding destination information. Session (RTP (Real-time Transport Protocol) / RTCP (Real-time Transport Control Protocol) session) establishment processing is executed.

  In the connection start process, a SIP request transmission / reception process (signaling process) according to a predetermined communication protocol (protocol) is executed via the server Sv. Specifically, according to the procedure of the SIP message, first, “INVITE (invitation)” is transmitted from the calling side (calling device T1) to the called side (calling device T2), and then the called side (calling device) When "100 Trying (provisional response)", "180 Ringing (calling)", "200 ok (success response)" is received from T2), the calling side (calling device T1) to the called side (calling device T2) “ACK (acknowledgment response)” is transmitted. This establishes a call connection (session establishment) between the calling party (calling apparatus T1) and the called party (calling apparatus T2).

  When the session is established, the calling side call device T1 is controlled so as to be able to start a call toward the call side call device T2, and thereafter, a predetermined call operation (for example, operation of a call button) is performed. Thus, a call can be made from the calling side to the called side. When a call is made between the call devices T1 and T2, a voice packet (RTP packet) obtained by packetizing voice data based on RTP / RTCP is transmitted / received, and the voice packet (RTP packet) is transmitted / received as necessary. Control packets (RTCP packets) for control are transmitted and received.

  Note that RTP is a protocol for transmitting and receiving audio data (or moving image data) itself, whereas RTCP periodically evaluates line quality such as packet loss, delay jitter, and round trip, This is a protocol for transmitting and receiving information in order to realize real-time communication corresponding to the band. Therefore, by transmitting / receiving the RTCP packet, for example, the state of the network and the states of the communication devices T1 and T2 are estimated from the information returned from the communication partner, and the communication state between the communication devices T1 and T2 is changed (communication). Dynamic processing such as reduction in the number of times) can be performed.

Here, the RTP / RTCP packet transmitted / received between the telephone apparatuses T1 and T2 after the connection start process is executed and the RTP / RTCP session is established will be described.
The RTP packet is composed of an IP header, a UDP (User Datagram Protocol) header, an RTP header, and RTP data. In this case, the RTP header includes a version information storage unit (V: version, for example, V = 2), a padding storage unit (P: padding), an extension bit storage unit (X: extension), and a CSRC (contributing source) from the top. Count storage unit (CC), marker information (M: marker) storage unit, marker bit storage unit (M: maker), payload type information storage unit (PT: payload type), sequence number information storage unit (sequence number), A time stamp storage unit (time stamp), an SSRC identifier storage unit (synchronization source identifier), and a CSRC identifier storage unit are provided. Then, behind the CSRC identifier storage section (when the extension header is used (when 1 is set in the extension bit storage section), behind the extension header section), between the telephone devices T1 and T2, specifically, the user P1 Therefore, audio data that is actually exchanged between P2 is added.

  On the other hand, the RTCP packet is composed of an IP header, a UDP header, an RTCP header, and RTCP data. In this case, the header of the RTCP APP packet, which is an extensible RTCP packet, includes a version information storage unit (V: version, for example, V = 2), a padding presence / absence storage unit (P: padding), a subtype storage unit ( So-called item count storage (IC: Item Count), packet type (PT) storage, report length storage (length), SSRC / CSRC identifier storage, ASCII (ASCII: American Standard Code for Information), information A Name storage unit described in (US standard code for exchange) is provided. A data storage unit (Application-Dependent Data) for storing application-specific data is added after the Name storage unit.

  In the packet type PT storage section, the type of the RTCP packet is recorded, and in this embodiment, as an example, 204 (packet type value) is set in the packet type (PT). The packet type value 204 indicates that the RTCP packet is APP (Application-defined: application-specific information), that is, a packet for notifying application-specific control information outside the RTCP standard.

  In addition to the APP packet (packet type value 204) as described above, the RTCP packet includes an RTCP SR (Sender Report) packet (packet type value 200) of the type sent from the RTP data sender, There are RTCP packets of the type sent from the recipients (RTCP RR (Receiver Report) packet (packet type value 201)).

  The SR packet is sent from a calling device that is sending a stream to another calling device. The SR packet is transmission information (sender info) that is information about a stream sent by the own device, and a stream for each received stream. A report block (reception report block) for reporting the reception state (packet discard rate, jitter, etc.) to the transmitting side communication device. On the other hand, the RR packet is for notifying information on a stream received from another call device, and a report block for reporting the reception status of the stream to the call device on the transmission side for each received stream. Contains.

  This report block includes the synchronization source (SSRC) identifier of the packet sender, the RTP loss rate, the number of lost RTP packets, the reception sequence number, the average jitter of the arrival time interval, and the last received SR. A transmission time (LSR: Last SR timestamp) and a time (DLSR: Delay since Last SR) from the time when the SR was last received until this RR is sent are set.

  Therefore, the transmitting side communication device (for example, the communication device T1) manages the number of transmitted RTP packets and the number of transmitted RTP bytes at the time of transmission of RTP data. The communication device T2) manages management information such as the number of received RTP packets, the number of lost RTP packets, and jitter of arrival time when receiving RTP data. As a result, the communication states can be grasped mutually in the communication devices T1 and T2.

By the way, in the IP telephone system according to the present embodiment, two communication data transmission modes (transmission interruption mode and transmission interrupt mode) are established between each of the communication devices T1 and T2 and the server Sv during a call between the communication devices T1 and T2. Transmission data (for example, audio data) is transmitted and received by arbitrarily switching (transmission restart mode).
That is, in the IP telephone system according to the present embodiment, the connection start process is executed, and the communication device T1, T2 and the server Sv are connected with each other in a state where the communication connection (call connection) between the communication devices T1, T2 is established. Control is performed to switch between a transmission interruption mode in which transmission of all communication data is interrupted and a transmission resumption mode in which transmission of the interrupted communication data is resumed by inputting a predetermined command. Transmission / reception data control means is provided.

  In this case, as shown in FIG. 2, the communication devices T1 and T2 are provided with a transmission / reception data control unit 27 as transmission / reception data control means, and the data control unit 27 includes a command for switching the transmission mode, specifically Specifically, the voice to the server Sv is input by inputting a command for switching from a state in which a call is being performed between the call devices T1 and T2 (hereinafter, this state is referred to as a standard call state) to a transmission interruption mode. Stop sending data. On the other hand, when the command for switching from the transmission interruption mode to the transmission resumption mode is input, the data control unit 27 resumes transmission of voice data to the server Sv suspended in the transmission interruption mode.

  On the other hand, the server Sv performs a notification command indicating that the transmission mode switching has been performed from the communication devices T1 and T2, as a transmission / reception data control means, specifically, switching from the standard call state to the transmission interruption mode. A transmission / reception data control function for stopping transmission of voice data to the communication devices T1 and T2 by inputting a notification command to the effect is added. On the other hand, the transmission / reception data control function of the server Sv responds to the communication devices T1 and T2 suspended in the transmission interruption mode by inputting a notification command for switching from the transmission interruption mode to the transmission resumption mode. Resume transmission of audio data.

  Hereinafter, an operation example of voice data transmission mode switching in the IP telephone system according to the present embodiment will be described with reference to FIG. In the description of the operation, a connection start process has already been executed between the call devices T1 and T2 with the call device T1 as the calling side and the call device T2 as the called side, and a call connection has been established between the call devices T1 and T2. It is assumed that there is an established state (an RTP / RTCP session is established), and in this state, control for switching the voice data transmission mode is performed between the telephone device T1 and the server Sv.

  After executing the connection start processing, the telephone device T1 adds (stores) voice data to the RTP packet and transmits the packet to the other party (calling side) telephone device T2 via the server Sv. (E1). In this case, the voice data includes voice data of a predetermined level or higher (hereinafter referred to as voiced data) exchanged at the time of a daily call, voice data of a predetermined level or lower, for example, silent data generated between conversations or user P1, Also included are noise data generated around P2 (hereinafter referred to as silence data).

Thereby, a call (conversation) can be performed between the call devices T1 and T2, that is, between the users P1 and P2. Normally, voice data is transmitted and received between the call devices T1 and T2 via the server Sv until the call (conversation) is completed. On the other hand, during a call between the users P1 and P2 (between the call devices T1 and T2), another event (for example, a visitor, a meal, a toilet, etc.) other than the call occurs, and the call cannot be continued as it is. In some cases, the situation must be interrupted.
In such a case, for example, when the user P1 deals with another event as described above while the telephone device T1 is in the standard call state, the silent data is digitized in the voice processing circuit 12 and added to the RTP packet during that period. Packets are continuously transmitted to Sv, resulting in a wasteful consumption of the battery (battery) of the communication device T1.

  Therefore, in the present embodiment, when it is necessary to interrupt a call once started as described above, the call device T1 interrupts packet transmission of all voice data (sound data and silence data). That is, by switching the call device T1 from the standard call state to the transmission interruption mode, the power consumption of the battery (battery) is suppressed and the power consumption of the call device T1 is reduced. In this case, in the communication device T1, the transmission / reception data control unit 27 constantly monitors the level (decibel value, volume, frequency, etc.) of the audio signal collected and output from the microphone 8, and the level is constant (hereinafter referred to as “the level”). When the voice data transmission mode drops below the reference level, a command (mode switching command) for switching the voice data transmission mode from the standard call state to the transmission interruption mode is automatically generated, and by executing the mode switching command, Packet transmission of voice data (sound data and silence data) is interrupted.

  The reference level is not particularly limited because it may be arbitrarily set according to the purpose of use of the IP telephone system and the usage environment. For example, the output level of useless voice data (silence data) collected from the microphone 8 It may be set as a threshold for determining the state (level level, output time, etc.). In this case, the reference level may be fixedly set in advance, or may be set arbitrarily (in stages) according to the call time or the remaining battery (battery) level at that time. .

  In addition, as described above, the generation / execution of the mode switching command is performed by constantly monitoring the level (decibel value, volume, frequency, etc.) of the audio signal that is transmitted and received by the transmission / reception data control unit 27 from the microphone 8, In addition to the configuration in which the output level of the audio signal falls below a threshold value (reference level) and it is determined whether or not a predetermined time has passed, it is performed by the user P1 instead of or in addition to this. The transmission / reception data control unit 27 is passively (manually) performed based on an explicit operation (for example, an operation of pressing a mode switching button or a speech operation of a keyword (meaningful word)). Also good.

  According to such a configuration, the user P1 can switch the voice data transmission mode from the standard call state to the transmission interruption mode according to the needs. As a result, the user P1 checks the battery (battery) remaining amount indicator displayed on the monitor (not marked in particular) of the communication device T1, and not only when the remaining amount is low but also sufficiently Even when there is a sufficient remaining amount, the mode switch command is generated and executed in anticipation of the call time in advance, and the voice data transmission mode is switched from the standard call state to the transmission interruption mode, thereby further reducing battery (battery) consumption. Can be suppressed.

  As described above, when the voice data transmission mode is switched from the standard call state to the transmission interruption mode in the call device T1, the call device T1 transmits an RTCP packet for notifying that the voice data transmission mode has been switched. It transmits toward the server Sv via the wireless LAN interface 18 (E2). At that time, the transmission / reception data control unit 27 sets the packet type value to 204 as an extended event of RTCP, and stores data notifying the switching from the standard call state to the transmission interruption mode in the data storage unit of the packet. An RTCP packet (APP packet) is generated.

  On the other hand, the server Sv that has received the RTCP packet notifying the voice data transmission mode switching from the call device T1 recognizes that the switch to the transmission interruption mode in the call device T1 has been performed (that is, the voice data transmission is interrupted). In this case, for example, when it is detected that the transmission of voice data from the communication device T1 is interrupted for a predetermined time, an RTCP packet for notifying that the interruption has been accepted is transmitted to the communication device T1 (E3). ).

  At that time, the server Sv sets the packet type value to 204 as an RTCP extended event, and stores RTCP in which data for notifying acceptance of switching from the standard call state to the transmission interruption mode is stored in the data storage unit of the packet. A packet (APP packet) is generated. In this case, the processing procedure for generating the RTCP packet (APP packet) in the server Sv (FIG. 4) is, for example, storing a program for executing the processing procedure in the ROM 44, and the stored program uses the RAM 46 as a work area. It is executed by the CPU 48. Instead of installing the program in the server Sv, for example, a circuit system for executing a similar processing procedure may be newly added to the server Sv.

  On the other hand, when the interruption of the voice data transmission cannot be recognized, the server Sv stores the RTCP packet (APP packet) storing the data for notifying the sequence number of the RTP packet to which the voice data received from the telephone device T1 is added (stored). ) And transmits the packet to the communication device T1. The call device T1 that has received such a packet resumes and continues the call with a part of the RTP packet missing. In this case, as an example, if the configuration is such that the call device T1 switches to the transmission interruption mode again, generates an RTCP packet (APP packet) to notify that, and retransmits it to the server Sv, It is possible to reliably switch to the transmission interruption mode. At this time, the transmission / reception data control unit 27 may perform the retry control (such as the number of retries and the interval) for switching to the transmission interruption mode.

The call device T1 that has received the RTCP packet for notifying the acceptance of the voice data transmission mode switching from the server Sv causes the transmission / reception data control unit 27 to stop RTP packetization of all voice data (sound data and silence data). The transmission of the packet is interrupted.
On the other hand, the server Sv that has transmitted the RTCP packet for notifying the acceptance of the switching of the voice data transmission mode to the call device T1 sets the voice data to be transmitted to the call device T1 as voice data (sound data) having a predetermined level or higher. Switch to a voice data transmission mode that interrupts transmission of voice data (silent data) below a predetermined level (hereinafter, this mode in the server Sv is also referred to as a transmission interruption mode). Transmit to device T1.

As a result, transmission of all voice data (sound data and silence data) from the call device T1 to the server Sv is interrupted, and not only the transmission processing when transmitting the voice data but also the server Sv to the call device T1. The transmission of the voice data (silent data) is interrupted, and the reception process when receiving the voice data transmitted by the voice data can be reduced. For this reason, the power consumption of the battery (battery) of the communication device T1 can be reduced very effectively, and the power consumption of the communication device T1 can be effectively reduced. As a result, it is possible to continuously use the communication device T1 for a long time.
Note that the effect of reducing the power consumption is not limited to the calling side call device T1, and the call side call device T2 also uses the voice data (sound data and silence) from the call device T1. Since the transmission of (data) is interrupted, the power consumption can be reduced.

  In the configuration described above, when the server Sv recognizes that the switching to the transmission interruption mode in the telephone device T1 has been performed, the server Sv interrupts only the transmission of voice data (silent data) below a predetermined level. The level of the voice data indicating whether or not to stop the transmission when the transmission interruption mode is set on the server Sv side is, for example, whether or not some meaningful voice is input depending on the purpose of use or the usage environment of the IP telephone system. Alternatively, it may be arbitrarily set based on criteria such as whether or not it has a voice level (decibel value of audio signal, volume and frequency, etc.) equivalent to everyday conversation, and is not particularly limited here. Further, in the transmission interruption mode, the transmission of all voice data (sound data and silent data) may be interrupted on the server Sv side as well as the communication device T1.

  As described above, according to the present embodiment, since the switching of the voice data transmission mode is performed by the notification by transmitting and receiving the RTCP packet (APP packet), the normal data stream (that is, the voice data processing sequence) The voice data transmission mode can be easily switched without being affected by the above. In addition, the receiving side (server Sv) that has received the notification determines whether or not it is necessary to switch the voice data transmission mode. For example, as a result of the determination, switching of the voice data transmission mode in the telephone device T1 is not permitted (not performed). It is also possible to have a configuration. Further, it is possible to ensure the continuity of the sequence number of the RTP packet to which the voice data is added (stored) and to easily confirm the sequence number.

Here, in the present embodiment, after switching from the standard call state to the transmission interruption mode, it is possible to switch (return) from the transmission interruption mode to the standard call state as necessary. The standard call state after switching from the transmission interruption mode is referred to as a transmission resumption mode.
When the communication device T1 is switched (returned) from the transmission interruption mode to the transmission resume mode, the communication device T1 transmits an RTCP packet for notifying that the voice data transmission mode has been switched (returned). It transmits toward the server Sv via the wireless LAN interface 18 (E5). At that time, the transmission / reception data control unit 27 sets the packet type value to 204 as an RTCP extended event, and also notifies the data storage unit of the packet of switching (returning) from the transmission suspension mode to the transmission resumption mode. RTCP packet (APP packet) is generated.
As a result, the communication device T1 resumes transmission of all voice data (sound data and silence data) to the server Sv.

  The mode switching (return) command is generated and executed by the transmission / reception data control unit 27 constantly monitoring the level (decibel value, volume, frequency, etc.) of the audio signal picked up from the microphone 8 and output. It may be automatically performed by determining whether or not a predetermined time has passed since the output level of the signal exceeds the threshold (reference level). In addition to or in addition to this, transmission / reception data control is performed based on an explicit operation performed by the user P1 (for example, an operation of pressing a mode switching button or a speech operation of a keyword (a meaningful word)). The unit 27 may be performed passively (manually).

  On the other hand, the server Sv that has received the RTCP packet for notifying the voice data transmission mode switching from the call device T1 recognizes that the switch to the transmission resume mode in the call device T1 has been performed (that is, the voice data transmission is resumed). In such a case, for example, when transmission of voice data from the call device T1 is detected within a predetermined time, an RTCP packet for notifying that the restart has been accepted is transmitted to the call device T1.

At that time, the server Sv sets the packet type value to 204 as an RTCP extended event, and stores RTCP in which data for notifying acceptance of switching from the transmission suspension mode to the transmission resumption mode is stored in the data storage unit of the packet. A packet (APP packet) is generated. In this case, the processing procedure for generating the RTCP packet (APP packet) in the server Sv (FIG. 4) is, for example, storing a program for executing the processing procedure in the ROM 44, and the stored program uses the RAM 46 as a work area. It is executed by the CPU 48. Instead of installing the program in the server Sv, for example, a circuit system for executing a similar processing procedure may be newly added to the server Sv.
Then, the server Sv that has transmitted the RTCP packet notifying the acceptance of the switching of the voice data transmission mode to the call device T1 not only the voice data (sound data) of a predetermined level or higher, but also the voice data (silent data) of a predetermined level or lower. Data) is also converted into RTP packets, and transmission of all voice data (sound data and silent data) to the communication device T1 is resumed.

Note that the switching from the standard call state to the transmission interruption mode and the switching from the transmission interruption mode to the transmission resumption mode can be controlled not only from the telephone device T1 but also from the server Sv. In this case, the server Sv transmits an RTCP packet for notifying the switching of the voice data transmission mode to the telephone device T1 (E4, E6, E7).
Specifically, when switching from the standard call state to the transmission interruption mode, the server Sv sets the packet type value to 204 as an RTCP extended event, and stores the packet from the standard call state in the data storage unit. An RTCP packet (APP packet) storing data for notifying switching to the transmission interruption mode is generated and transmitted to the communication device T1 (E4, E7). On the other hand, when switching from the transmission interruption mode to the transmission resumption mode, the server Sv sets the packet type value to 204 as an RTCP extended event and transmits the packet from the transmission interruption mode to the data storage unit of the packet. An RTCP packet (APP packet) storing data for notifying switching to the resume mode is generated and transmitted to the communication device T1 (E6).

  That is, the server Sv can voluntarily switch from the standard call state to the transmission interruption mode and notify the call device T1 to that effect (E6). When there is a need to transmit (sound data) (for example, when making a simultaneous report in an emergency), it is possible to switch (return) from the transmission interruption mode to the transmission resumption mode and notify that fact. (E7). Thereby, it is possible to freely switch between the standard call state (transmission resume mode) and the transmission interruption mode between the telephone device T1 and the server Sv as necessary.

  In the above-described embodiment, audio data at a predetermined level or higher is defined as sound data, and audio data at a predetermined level or lower is defined as silence data. However, the present invention is not limited to this, and is based on other criteria. The level of audio data may be set. For example, when voice data includes a predetermined amount of voice data, this is set as voice data at a predetermined level or higher. On the other hand, when voice data includes a predetermined amount of silence data, the voice data is set at a predetermined level or lower. It may be defined as data.

  In this case, in the server Sv, it is necessary to determine whether or not such a criterion is satisfied. For example, a determination function is further added to the above-described transmission / reception data control function, and the determination process is executed by the determination function. The voice data transmission mode may be switched based on the determination result, or a determination unit (not shown) is separately constructed, and the determination process is executed by the determination unit. The data transmission mode may be switched.

  Further, as a reference for switching the voice data transmission mode, the radio field intensity measurement data acquired by the radio field intensity measurement unit 20 of each relay apparatus 1AP, 2AP, the positions of the communication apparatuses T1, T2 specified based on the data, etc. , And based on this, the voice data transmission mode between the telephone devices T1, T2 and the server Sv may be switched.

  With such switching criteria, it is possible to freely switch between the standard call state (transmission resume mode) and the transmission suspension mode, so that the standard call state (transmission resume mode) or the transmission suspension mode can be arbitrarily selected. In addition, not only can the power consumption of the call device T1 be reduced, but the call quality can be freely changed according to the call environment. For example, in places where there is a lot of noise in the surroundings, as a transmission interruption mode, the battery (battery) is prevented from being consumed, while moving to a quiet place and switching to the standard call state (transmission resume mode) after moving, call quality It becomes possible to continue the call in a state where the value is further increased.

BRIEF DESCRIPTION OF THE DRAWINGS It is a figure which shows the structure of the IP telephone system which concerns on one embodiment of this invention, Comprising: (a) is the whole system outline figure which shows the state in which the telephone call is performed between telephone apparatus, (b) is a voice The sequence diagram at the time of switching of a data transmission mode. The figure which shows the internal structure of a communication apparatus. The figure which shows the internal structure of a relay apparatus. The figure which shows the internal structure of a server.

Explanation of symbols

T1, T2 call device 6 server

Claims (3)

A plurality of communication terminals capable of communicating with each other via a network;
An IP having a server for executing connection start processing when communicating between the communication terminals and controlling communication data transmitted and received between the communication terminals based on RTP during communication between the communication terminals A telephone system,
In the state where the connection start process is executed, the communication connection between the communication terminals is established, and the communication data converted into RTP packets is transmitted and received between the communication terminal and the server.
A transmission / reception data control means for performing a control for switching between a transmission interruption mode in which transmission of the communication data is interrupted and a transmission resumption mode in which transmission of the communication data is resumed when a predetermined command is input; An IP telephone system characterized by The communication terminal controls the server to switch between the transmission interruption mode and the transmission resumption mode of the communication data by inputting a command for switching the mode as the transmission / reception data control means. A transmission / reception data control unit is provided.
The server receives, as the transmission / reception data control means, a notification command indicating that the mode switching of the communication terminal has been performed, whereby the communication data transmission interruption mode and the communication data are transmitted to the communication terminal. 2. The IP telephone system according to claim 1, further comprising a transmission / reception data control function for performing control for switching between transmission resumption modes.   The IP telephone system according to claim 2, wherein the notification command is transmitted from the communication terminal to the server as an RTCP extended event.

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